This topic describes the parameters and interface details of the Fun-ASR-Flash non-real-time speech recognition HTTP API.
For input requirements such as supported audio formats, file size limits, and duration limits, see Audio specifications.
In streaming mode, the client needs to handle the following:
Service endpoints
- Singapore
- China (Beijing)
POST https://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api/v1/services/aigc/multimodal-generation/generationReplace {WorkspaceId} with your actual Workspace ID.Request headers
| Parameter | Type | Required | Description |
|---|---|---|---|
Authorization | string | Yes | The authentication token, in the format Bearer <your_api_key>. Replace "<your_api_key>" with your actual API key. |
Content-Type | string | Yes | The media type of the request body. Fixed to application/json. |
X-DashScope-SSE | string | No | Controls whether results are returned as an SSE stream. Set to enable to turn on SSE streaming. The server returns intermediate and final recognition results in multiple messages only for audio that is at least 1 minute long. Set to disable or omit the parameter to return only the final result. |
Request bodymodelstring(Required)The model name.inputobject(Required)The input information.
Properties messages array(object)(Required)The message list. It contains the audio to recognize and, optionally, conversation context that improves recognition accuracy.
Properties role string(Required)The message role. Valid values:
array(object)(Required)The message content list.
Properties type string(Required)The content type. Each request needs at least one message of type input_audio. Valid values:
object(Conditionally required)Required when type is input_audio.
Properties data string(Required)The audio data to recognize. For input requirements such as supported audio formats, file size limits, and duration limits, see Audio specifications. Two methods are supported:
https://example.com/audio/sample.wavExample (Base64): data:audio/wav;base64,{BASE64_ENCODED_DATA}string(Conditionally required)When type is input_text, enter the recognition results of the user's speech from previous turns or a domain-specific word list. When type is text, enter the large language model's replies from previous turns. Text length is measured in characters, where each character counts as 1. The combined length of the text fields across all messages in a single context turn cannot exceed 400 characters. Any excess is truncated from the end.object(Required)The model parameters.Text polishing is disabled by default and is not yet available.Text polishing: While transcribing speech, the model removes meaningless filler words and stuttered repetitions, handles self-corrections, smooths out colloquial expressions, and standardizes punctuation and text formatting. The output is more concise, fluent, and readable while preserving the speaker’s final intent and key information as much as possible.
Properties format string(Required)The audio format. Set this to match your actual audio format. Supported values include wav, mp3, and opus. For details, see Audio specifications.sample_ratestring(Optional)The audio sample rate, in Hz. For example, 16000 means a 16 kHz sample rate. For details, see Audio specifications.vocabulary_idstring(Optional)The ID of a precompiled hot word list.Generate this ID in advance by calling the create hot word list API. Pass the ID during recognition to use the hot words in the list.Suitable for scenarios where the vocabulary is known and relatively stable, and where you need to reuse the same word list across requests.For usage details, see Precompiled hotwords.language_hints array[string](Optional)The language codes to recognize. If the language is unknown in advance, omit this parameter to let the model detect it automatically. You can set up to 1 value. If you set more, only the first 1 take effect.
Supported language codes
| The following examples use the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration differs across regions, and the API key for the Singapore region differs from the one for the Beijing region.
|
Response bodyrequest_idstringThe unique identifier of this request.outputobjectThe output result.
Properties text stringThe full recognized text accumulated so far.sentenceobjectThe details of the current sentence.
Properties sentence_id integerThe sentence number, starting from 1.sentence_endbooleanWhether this is the final result for the sentence. true indicates that recognition of the sentence is complete.begin_timeintegerThe start time of the sentence, in milliseconds.end_timeintegerThe end time of the sentence, in milliseconds. Returned only when sentence_end is true.textstringThe recognized text of the current sentence.channel_idintegerThe channel number, starting from 0.wordsarrayThe list of word-level timestamps.
Properties text stringThe word text.begin_timeintegerThe start time of the word, in milliseconds.end_timeintegerThe end time of the word, in milliseconds.punctuationstringThe punctuation mark after the word. An empty string when there is no punctuation.fixedbooleanWhether the word is stabilized. false indicates that the word's timestamp may be adjusted in subsequent events.objectThe usage information. Returned only when sentence_end is true.
Properties duration integerThe processed audio duration, in seconds. |
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SSE streaming result processing logic
In streaming mode, the client needs to handle the following:
- For each SSE event received, parse the JSON in the
datafield. - Use
output.sentence.sentence_endto determine whether the current sentence has ended. When it istrue, recognition is complete, the word-level timestamps are stabilized, and the result can be used as final. When it isfalse, recognition is still in progress, and text and timestamps may be updated in subsequent events. - The
usageinformation is returned only in sentence-end events and includes the processed audio duration.